[asterisk-users] Outgoing PSTN calls , unusable voice quality

Sun Dec 2 08:51:41 CST 2007

Dear Salvatore and Joanna,
Thank you much for both your detailed explanations.
I will surely check my firewall configuration and logs to make sure the 
VoIP traffic is passing correctly.
However, I'm a bit confused as the problems that I'm experiencing are 
with calls made via the
Sangoma analog card.
So the voice goes from the SIP phone through asterisk through the 
sangoma card then
directly into the PSTN and vice versa. There are no firewalls in the way.
Furthermore incoming calls are OK, the problem is only with outgoing 
calls when I can hear the other party well
but they barely understand me.

Is there any major difference in the way that Incoming/Outgoing calls 
are processed in the above scenario?
Any way that I could trace those processes for faults?

Thank you again.

Regards,
Veselin

Salvatore Giudice wrote:
> When you take your packet capture, you'll need to look at the sip messages
> with SDP attached to get the ip's and ports used for both media streams.
> Make sure that the ips are correct and that the port used can traverse
> between those ip's without being blocked by a packet filter or firewall. A
> lot of times, administrators will set a range of UDP ports that are allowed
> to pass their packet filter for media and your pbx or phones may be using a
> different range. This can cause audio loss. You'll need to eliminate that
> possibility. Sometimes checking your firewall/packet filters for blocks may
> also prove helpful in identifying problems. You should be aware that the
> logs from certain firewall products may not be comprehensive. For example,
> in the past I have seen packets dropped going through netscreens because of
> invalid headers and no entries appeared in the logs. If you ultimately
> believe a firewall may be blocking your traffic make sure you setup a
> capture port or a span on each side of the device and verify the traffic
> going to and leaving from the firewall using ethereal on a laptop or maybe a
> Nixon box if you are in a large distributed environment. Never trust a
> potentially broken device to report accurate information about it's
> function.
>
> TDM = Time Division Multiplexing
>
> TDM describes how channels are separated on T1's, etc. It's common to refer
> to those types of connections as TDM.
> http://en.wikipedia.org/wiki/Time-division_multiplexing
>
>
> --------------------------------------------------
> Salvatore Giudice
> Salvatore.Giudice at VoIPSecurityTraining.com
>
> VoIP Security Training, LLC
> http://VoIPSecurityTraining.com
>
> 848 N. Rainbow Blvd. #1676
> Las Vegas, NV 89107
> Phone: (617) 959-7625
> Fax: (214) 279-2906
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Veselin
> Kantsev
> Sent: Friday, November 30, 2007 8:28 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality
>
> Thank you much for the prompt reply Salvatore.
>
> Would you have the time to explain further how should I go for verifying
> that SDP and RTP are OK.
> Also what is reffered to as the TDM site.
>
> Veselin
>
> On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote:
>   
>> Take a packet capture of your VoIP segment and verify that the SDP is
>> correct and that the RTP is making it to the correct places. If all that
>> looks good and this is a straight out quality problem, then you need to
>> figure out if it's happening on the voip side or on the TDM side. You
>>     
> should
>   
>> make calls (with captures) VoIP to Voip passing the media through your
>> asterisk and also try routing a tdm call in and back out. If you have the
>> equipment, take a mos score of the TDM loop.
>>
>> Without any of the above, you will not be able to isolate the issue.
>>
>> --------------------------------------------------
>> Salvatore Giudice
>> Salvatore.Giudice at VoIPSecurityTraining.com
>>
>> VoIP Security Training, LLC
>> http://VoIPSecurityTraining.com
>>
>> 848 N. Rainbow Blvd. #1676
>> Las Vegas, NV 89107
>> Phone: (617) 959-7625
>> Fax: (214) 279-2906
>>
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Veselin
>> Kantsev
>> Sent: Friday, November 30, 2007 2:47 PM
>> To: asterisk-users at lists.digium.com
>> Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality
>>
>> Hello,
>> I have an Asterisk running with a Sangoma A200 card with Hardware Echo 
>> cancelling connected to the UK PSTN.
>> If a PSTN call comes in, voice both ways is OK, however if an outgoing 
>> call over the PSTN is made I can hear the other party OK but they can 
>> not, they can barely understand what I am saying, my voice is unclear 
>> fading and skipping.
>> Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 
>> are OK too. I've tried gsm/ulaw/alaw codecs so far.
>> Tried disabling the echo cancelling as well.
>>
>> Any suggestions will be greatly appreciated.
>>
>>
>> Regards,
>> Veselin
>>
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>
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-- 
Regards,
Veselin Kantsev
Campbell-Lange Workshop

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